Re: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with same Gateway

So there seems to be some sort of error when bridging directly like originate {ignore_early_media=true}sofia/gateway/.com/91415992 &bridge(sofia/gateway/.com/91415465 ) BUT if I get FS to send media to leg A, and then

[Freeswitch-users] can't pass full sip url to dialplan

Hi: I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is : *X-lite(softphone) dials -> 1...@4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receive

Re: [Freeswitch-users] MFC-R2 support for FreeSWITCH

Way to go moy! On Fri, Aug 21, 2009 at 7:59 PM, Michael Collins wrote: > > > On Fri, Aug 21, 2009 at 3:29 PM, Moises Silva wrote: > >> So, I finally took some days to put up OpenR2 working with OpenZAP, which >> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has >> support for

Re: [Freeswitch-users] MFC-R2 support for FreeSWITCH

On Fri, Aug 21, 2009 at 3:29 PM, Moises Silva wrote: > So, I finally took some days to put up OpenR2 working with OpenZAP, which > means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has > support for. Including Mexico, Brazil, Argentina and others. The stack has > been used by Aster

[Freeswitch-users] MFC-R2 support for FreeSWITCH

So, I finally took some days to put up OpenR2 working with OpenZAP, which means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has support for. Including Mexico, Brazil, Argentina and others. The stack has been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most coun

Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

probably disk i/o. Is it some kind of flash drive? make a ramdisk and simlink in /usr/local/freeswitch/db and /usr/local/freeswitch/log to it the default configuration uses a lot of high level features that use the sqlite db on the disk. We also offer commercial support where we could dig deeper

Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

On Fri, Aug 21, 2009 at 04:15:13PM -0300, Rogelio Perez wrote: > Hi Everyone, > > I'm working on a PBX project for the Sheevaplug ARM based computer, > with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU. > So far I've found a big difference between Freeswitch and Asterisk > performance

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

try setting FS to 30ms too edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it looks like p...@30i from: to: On Fri, Aug 21, 2009 at 1:38 PM, Brian West wrote: > You can ship me one whois bkw.org, I can add it to my lab. > > /b > > On Aug 21, 2009, at 10:38 AM,

[Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

Hi Everyone, I'm working on a PBX project for the Sheevaplug ARM based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU. So far I've found a big difference between Freeswitch and Asterisk performance times. This is a comparison of the time it takes them to perform different

Re: [Freeswitch-users] Loopback and bypass_media

Hi there, I created a feature request to cover this issue: http://jira.freeswitch.org/browse/FSCORE-422 - The ability to support Call FollowMe (or Call Blast) and multiple termination carriers - without Loopback If anybody wants comment on its merits and/or make the request clearer - that would

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

You can ship me one whois bkw.org, I can add it to my lab. /b On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote: > > BTW: We can ship you a FritzBox if you need one for testing. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org ht

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

Hello Mathieu, thank for your help. But this however didn't change the behaviour. I've read of a patch in mod_sofia.c which partly corrects the problem temporarily: When I change Line 784 to if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms) { to if (switch_rtp_rea

Re: [Freeswitch-users] hanging problem with switch_scheduler_add_task

Use gcore to get a core dump. It will pause the process for the duration of the dump, but will not kill the process. On Fri, Aug 21, 2009 at 11:24 AM, mark morreny wrote: > Hi Mathieu, > > Thanks for your help. > > How can I intentionally crash a hanging freeswitch and obtain the core file > to r

Re: [Freeswitch-users] zombie channels

Hello, > I tried running the core, but I am getting some errors: >  ./freeswitch-gcore > /usr/local/freeswitch/log/freeswitch.gcore.fm5478:1: Error in sourced > command file: > ptrace: No such process. > gcore: failed to create /usr/local/freeswitch/log/freeswitch.gcore.16240 > > What is the prope

Re: [Freeswitch-users] zombie channels

Hello, Yes, I am using cdr, so I guess CS_REPORTING could be a problem. I tried running the core, but I am getting some errors: ./freeswitch-gcore /usr/local/freeswitch/log/freeswitch.gcore.fm5478:1: Error in sourced command file: ptrace: No such process. gcore: failed to create /usr/local/frees

Re: [Freeswitch-users] hanging problem with switch_scheduler_add_task

Hi Mathieu, Thanks for your help. How can I intentionally crash a hanging freeswitch and obtain the core file to run gdb? Thanks, Mark On Fri, Aug 21, 2009 at 11:59 PM, Mathieu Rene wrote: > Means another task is hanging, do a "thread apply all bt" in gdb and > look for scheduler tasks. > > M

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

Try setting that in your sip profile: Thats a feature to work around with devices lying about their ptime in their sdp payload. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:

Re: [Freeswitch-users] hanging problem with switch_scheduler_add_task

Means another task is hanging, do a "thread apply all bt" in gdb and look for scheduler tasks. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 21-Aug-09, at 11:41 AM, mark morreny wrote: > Hi, > > I am experiencing some ha

Re: [Freeswitch-users] Sangoma A500 and FreeSWITCH

I already tried the boostbri config, but with no success... I will contact the support. Thank you! Best regards, V. Panayotov ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch

[Freeswitch-users] hanging problem with switch_scheduler_add_task

Hi, I am experiencing some hanging when fs is executing switch_scheduler_add_task. switch_scheduler_add_task(switch_epoch_time_now(NULL) , data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL); I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION and sometimes, ha

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

Hello Michael, I made some tests with Freeswitch and Fritzbox and found by Wireshark that: within one call * Freeswitch starts sending 20msec packets, then after ~0,2 second sends 30msec packets * FritzBox always sends 30msec packets. The average jitter is below 2 msec in both dire

Re: [Freeswitch-users] Application Variable list needed

Be sure to visit wiki.freeswitch.org and search for "rosetta stone" which will take you to a page that helps translate Asterisk concepts into FreeSWITCH concepts. It will seem strange at first but once you get over the hump you will really appreciate the power of FreeSWITCH. Be sure to visi

Re: [Freeswitch-users] Sangoma A500 and FreeSWITCH

> > Can you give me some pointers about A500 configuration with FreeSWITCH? I > need nothing fancy - just inbound/outbound calls to telco. > There is some progress and you should be able to get it working with this: http://wiki.sangoma.com/boostbri If you find any problem please contact techd...@

Re: [Freeswitch-users] zombie channels

Hi, CS_REPORTING is the state in which cdrs are written, if the channel gets stuck in that state, the cdr module you are using is probably hanging somewhere. Use the "freeswitch-gcore" script in your source tree's scripts directory to generate a bug report for hanging channels. should be l

Re: [Freeswitch-users] Call exits after 120 seconds with hangup cause

Do you have those lines in switch.conf file? BR ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/option

Re: [Freeswitch-users] Call exits after 120 seconds with hangup cause

You have a NAT issue. /b On Aug 21, 2009, at 7:46 AM, Dennis wrote: > hi, > > we are using fs for different services, but we never used it, to > connect sip-phones directly with fs. > > now we want to do so, but we encounter big problems. everything works > fine, but after 120 seconds fs hangs u

Re: [Freeswitch-users] Accepting google talk friend requests

Hi Tapan, if your google talk is loaded as client mode in FreeSWITCH, you cannot automatically accept new incoming invite requests. But if you have google talk (mod_dingaling) loaded as component mode (assuming you have your own jabber server setup properly with google federation etc, and loaded w

[Freeswitch-users] Call exits after 120 seconds with hangup cause

hi, we are using fs for different services, but we never used it, to connect sip-phones directly with fs. now we want to do so, but we encounter big problems. everything works fine, but after 120 seconds fs hangs up with the hangup cause RECOVERY_ON_TIMER_EXPIRE. it seems that this has something

[Freeswitch-users] Sangoma A500 and FreeSWITCH

Hi, Is it already possible to use FreeSWITCH with A500 BRI card? I found a thread ( http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg09633.html) stating that the support is not quite ready. Can you give me some pointers about A500 configuration with FreeSWITCH? I need nothing

[Freeswitch-users] Accepting google talk friend requests

Hi Folks - Im a relative newbie freeswitch and first of all wanted to thank you for all the great work u have done here. My question is about mod dingaling, and specifically being able to get incoming calls from Google Talk. Ive got the client set up on Freeswitch, and am able to receive calls,

Re: [Freeswitch-users] zombie channels

Hi Woody, 2009/8/21 Woody Dickson : > After a high traffic session, I do "show channels", I would find a bunch of > "CS_HIBERNATE" channels that don't get removed after all the traffic is > gone. > > Does anyone know what is the case of thoes CS_HIBERNATE'd channels?  How can > I set  a timeout f

Re: [Freeswitch-users] Choppy voive in conference from fritzbox to freeswitch (ptime:30 msecs)

On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote: > Just curious - if it seems to be working with Asterisk but not > FreeSWITCH then could you do some tcpdumps of working vs. non-working > calls and then analyze them with Wireshark? I think Jason Garland's > ClueCon presentation(s) might

[Freeswitch-users] zombie channels

Hi, I am running 1.0.4 right now using latest trunk. After a high traffic session, I do "show channels", I would find a bunch of "CS_HIBERNATE" channels that don't get removed after all the traffic is gone. Does anyone know what is the case of thoes CS_HIBERNATE'd channels? How can I set a tim

[Freeswitch-users] Dialing SIP URL issue

Hi: I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is : *X-lite(softphone) dials -> 1...@4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receive